/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2005 Dave Chapman * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version 2 * of the License, or (at your option) any later version. * * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ #include "codeclib.h" #include "libm4a/m4a.h" #include "libfaad/common.h" #include "libfaad/structs.h" #include "libfaad/decoder.h" CODEC_HEADER /* this is the codec entry point */ enum codec_status codec_main(void) { /* Note that when dealing with QuickTime/MPEG4 files, terminology is * a bit confusing. Files with sound are split up in chunks, where * each chunk contains one or more samples. Each sample in turn * contains a number of "sound samples" (the kind you refer to with * the sampling frequency). */ size_t n; static demux_res_t demux_res; stream_t input_stream; uint32_t sound_samples_done; uint32_t elapsed_time; uint32_t sample_duration; uint32_t sample_byte_size; int file_offset; int framelength; int lead_trim = 0; unsigned int i; unsigned char* buffer; static NeAACDecFrameInfo frame_info; NeAACDecHandle decoder; int err; uint32_t s = 0; unsigned char c = 0; /* Generic codec initialisation */ ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512); ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED); ci->configure(DSP_SET_SAMPLE_DEPTH, 29); next_track: err = CODEC_OK; if (codec_init()) { LOGF("FAAD: Codec init error\n"); err = CODEC_ERROR; goto exit; } while (!*ci->taginfo_ready && !ci->stop_codec) ci->sleep(1); sound_samples_done = ci->id3->offset; ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); codec_set_replaygain(ci->id3); stream_create(&input_stream,ci); /* if qtmovie_read returns successfully, the stream is up to * the movie data, which can be used directly by the decoder */ if (!qtmovie_read(&input_stream, &demux_res)) { LOGF("FAAD: File init error\n"); err = CODEC_ERROR; goto done; } /* initialise the sound converter */ decoder = NeAACDecOpen(); if (!decoder) { LOGF("FAAD: Decode open error\n"); err = CODEC_ERROR; goto done; } NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder); conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */ NeAACDecSetConfiguration(decoder, conf); err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c); if (err) { LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type); err = CODEC_ERROR; goto done; } ci->id3->frequency = s; i = 0; if (sound_samples_done > 0) { if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done, &sound_samples_done, (int*) &i)) { elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100); ci->set_elapsed(elapsed_time); } else { sound_samples_done = 0; } } if (i == 0) { lead_trim = ci->id3->lead_trim; } /* The main decoding loop */ while (i < demux_res.num_sample_byte_sizes) { ci->yield(); if (ci->stop_codec || ci->new_track) { break; } /* Deal with any pending seek requests */ if (ci->seek_time) { if (alac_seek(&demux_res, &input_stream, ((ci->seek_time-1)/10)*(ci->id3->frequency/100), &sound_samples_done, (int*) &i)) { elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100); ci->set_elapsed(elapsed_time); if (i == 0) { lead_trim = ci->id3->lead_trim; } } ci->seek_complete(); } /* Lookup the length (in samples and bytes) of block i */ if (!get_sample_info(&demux_res, i, &sample_duration, &sample_byte_size)) { LOGF("AAC: get_sample_info error\n"); err = CODEC_ERROR; goto done; } /* There can be gaps between chunks, so skip ahead if needed. It * doesn't seem to happen much, but it probably means that a * "proper" file can have chunks out of order. Why one would want * that an good question (but files with gaps do exist, so who * knows?), so we don't support that - for now, at least. */ file_offset = get_sample_offset(&demux_res, i); if (file_offset > ci->curpos) { ci->advance_buffer(file_offset - ci->curpos); } else if (file_offset == 0) { LOGF("AAC: get_sample_offset error\n"); err = CODEC_ERROR; goto done; } /* Request the required number of bytes from the input buffer */ buffer=ci->request_buffer(&n,sample_byte_size); /* Decode one block - returned samples will be host-endian */ NeAACDecDecode(decoder, &frame_info, buffer, n); /* Ignore return value, we access samples in the decoder struct * directly. */ if (frame_info.error > 0) { LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error)); err = CODEC_ERROR; goto done; } /* Advance codec buffer */ ci->advance_buffer(n); /* Output the audio */ ci->yield(); framelength = (frame_info.samples >> 1) - lead_trim; if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0) { /* Currently limited to at most one frame of tail_trim. * Seems to be enough. */ if (ci->id3->tail_trim == 0 && sample_duration < (frame_info.samples >> 1)) { /* Subtract lead_trim just in case we decode a file with * only one audio frame with actual data. */ framelength = sample_duration - lead_trim; } else { framelength -= ci->id3->tail_trim; } } if (framelength > 0) { ci->pcmbuf_insert(&decoder->time_out[0][lead_trim], &decoder->time_out[1][lead_trim], framelength); } if (lead_trim > 0) { /* frame_info.samples can be 0 for the first frame */ lead_trim -= (i > 0 || frame_info.samples) ? (frame_info.samples >> 1) : sample_duration; if (lead_trim < 0 || ci->id3->lead_trim == 0) { lead_trim = 0; } } /* Update the elapsed-time indicator */ sound_samples_done += sample_duration; elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100); ci->set_elapsed(elapsed_time); /* Keep track of current position - for resuming */ ci->set_offset(elapsed_time); i++; } err = CODEC_OK; done: LOGF("AAC: Decoded %lu samples\n", sound_samples_done); if (ci->request_next_track()) goto next_track; exit: return err; }